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NetBSD/sys/dev/mulaw.c

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merge ktrace-lwp.
2005-12-11 15:16:03 +03:00
/* $NetBSD: mulaw.c,v 1.24 2005/12/11 12:20:53 christos Exp $ */
RCS ID police.
1997-06-15 02:25:11 +04:00
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
/*
* Copyright (c) 1991-1993 Regents of the University of California.
* All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. All advertising materials mentioning features or use of this software
* must display the following acknowledgement:
* This product includes software developed by the Computer Systems
* Engineering Group at Lawrence Berkeley Laboratory.
* 4. Neither the name of the University nor of the Laboratory may be used
* to endorse or promote products derived from this software without
* specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND
* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
* ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE
* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
* SUCH DAMAGE.
*
*/
add RCSIDs
2001-11-13 08:32:49 +03:00
#include <sys/cdefs.h>
merge ktrace-lwp.
2005-12-11 15:16:03 +03:00
__KERNEL_RCSID(0, "$NetBSD: mulaw.c,v 1.24 2005/12/11 12:20:53 christos Exp $");
add RCSIDs
2001-11-13 08:32:49 +03:00
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
#include <sys/types.h>
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
#include <sys/systm.h>
#include <dev/auconv.h>
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
#include <dev/mulaw.h>
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
/* #define MULAW_DEBUG */
#ifdef MULAW_DEBUG
# define DPRINTF(x) printf x
#else
# define DPRINTF(x)
#endif
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
/*
Respect the law: Use A-law and mu-law as spellings as far as easily possible. Inspired by Igor Sobrado in PR 19680.
2003-04-06 22:20:07 +04:00
* This table converts a (8 bit) mu-law value to a 16 bit value.
fix typo in comment
2000-08-18 09:55:31 +04:00
* The 16 bits are represented as an array of two bytes for easier access
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
* to the individual bytes.
*/
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t mulawtolin16[256][2] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
{0x02,0x84}, {0x06,0x84}, {0x0a,0x84}, {0x0e,0x84},
{0x12,0x84}, {0x16,0x84}, {0x1a,0x84}, {0x1e,0x84},
{0x22,0x84}, {0x26,0x84}, {0x2a,0x84}, {0x2e,0x84},
{0x32,0x84}, {0x36,0x84}, {0x3a,0x84}, {0x3e,0x84},
{0x41,0x84}, {0x43,0x84}, {0x45,0x84}, {0x47,0x84},
{0x49,0x84}, {0x4b,0x84}, {0x4d,0x84}, {0x4f,0x84},
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{0x7c,0x94}, {0x7c,0xb4}, {0x7c,0xd4}, {0x7c,0xf4},
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{0x7d,0x94}, {0x7d,0xb4}, {0x7d,0xd4}, {0x7d,0xf4},
{0x7e,0x14}, {0x7e,0x34}, {0x7e,0x54}, {0x7e,0x74},
{0x7e,0x8c}, {0x7e,0x9c}, {0x7e,0xac}, {0x7e,0xbc},
{0x7e,0xcc}, {0x7e,0xdc}, {0x7e,0xec}, {0x7e,0xfc},
{0x7f,0x0c}, {0x7f,0x1c}, {0x7f,0x2c}, {0x7f,0x3c},
{0x7f,0x4c}, {0x7f,0x5c}, {0x7f,0x6c}, {0x7f,0x7c},
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{0x7f,0xc8}, {0x7f,0xd0}, {0x7f,0xd8}, {0x7f,0xe0},
{0x7f,0xe8}, {0x7f,0xf0}, {0x7f,0xf8}, {0x80,0x00},
{0xfd,0x7c}, {0xf9,0x7c}, {0xf5,0x7c}, {0xf1,0x7c},
{0xed,0x7c}, {0xe9,0x7c}, {0xe5,0x7c}, {0xe1,0x7c},
{0xdd,0x7c}, {0xd9,0x7c}, {0xd5,0x7c}, {0xd1,0x7c},
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{0xbe,0x7c}, {0xbc,0x7c}, {0xba,0x7c}, {0xb8,0x7c},
{0xb6,0x7c}, {0xb4,0x7c}, {0xb2,0x7c}, {0xb0,0x7c},
{0xae,0x7c}, {0xac,0x7c}, {0xaa,0x7c}, {0xa8,0x7c},
{0xa6,0x7c}, {0xa4,0x7c}, {0xa2,0x7c}, {0xa0,0x7c},
{0x9e,0xfc}, {0x9d,0xfc}, {0x9c,0xfc}, {0x9b,0xfc},
{0x9a,0xfc}, {0x99,0xfc}, {0x98,0xfc}, {0x97,0xfc},
{0x96,0xfc}, {0x95,0xfc}, {0x94,0xfc}, {0x93,0xfc},
{0x92,0xfc}, {0x91,0xfc}, {0x90,0xfc}, {0x8f,0xfc},
{0x8f,0x3c}, {0x8e,0xbc}, {0x8e,0x3c}, {0x8d,0xbc},
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{0x89,0x3c}, {0x88,0xbc}, {0x88,0x3c}, {0x87,0xbc},
{0x87,0x5c}, {0x87,0x1c}, {0x86,0xdc}, {0x86,0x9c},
{0x86,0x5c}, {0x86,0x1c}, {0x85,0xdc}, {0x85,0x9c},
{0x85,0x5c}, {0x85,0x1c}, {0x84,0xdc}, {0x84,0x9c},
{0x84,0x5c}, {0x84,0x1c}, {0x83,0xdc}, {0x83,0x9c},
{0x83,0x6c}, {0x83,0x4c}, {0x83,0x2c}, {0x83,0x0c},
{0x82,0xec}, {0x82,0xcc}, {0x82,0xac}, {0x82,0x8c},
{0x82,0x6c}, {0x82,0x4c}, {0x82,0x2c}, {0x82,0x0c},
{0x81,0xec}, {0x81,0xcc}, {0x81,0xac}, {0x81,0x8c},
{0x81,0x74}, {0x81,0x64}, {0x81,0x54}, {0x81,0x44},
{0x81,0x34}, {0x81,0x24}, {0x81,0x14}, {0x81,0x04},
{0x80,0xf4}, {0x80,0xe4}, {0x80,0xd4}, {0x80,0xc4},
{0x80,0xb4}, {0x80,0xa4}, {0x80,0x94}, {0x80,0x84},
{0x80,0x78}, {0x80,0x70}, {0x80,0x68}, {0x80,0x60},
{0x80,0x58}, {0x80,0x50}, {0x80,0x48}, {0x80,0x40},
{0x80,0x38}, {0x80,0x30}, {0x80,0x28}, {0x80,0x20},
{0x80,0x18}, {0x80,0x10}, {0x80,0x08}, {0x80,0x00},
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t lintomulaw[256] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x01,
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0x81, 0x81, 0x81, 0x81, 0x80, 0x80, 0x80, 0x80,
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t alawtolin16[256][2] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
{0x6a,0x80}, {0x6b,0x80}, {0x68,0x80}, {0x69,0x80},
{0x6e,0x80}, {0x6f,0x80}, {0x6c,0x80}, {0x6d,0x80},
{0x62,0x80}, {0x63,0x80}, {0x60,0x80}, {0x61,0x80},
{0x66,0x80}, {0x67,0x80}, {0x64,0x80}, {0x65,0x80},
{0x75,0x40}, {0x75,0xc0}, {0x74,0x40}, {0x74,0xc0},
{0x77,0x40}, {0x77,0xc0}, {0x76,0x40}, {0x76,0xc0},
{0x71,0x40}, {0x71,0xc0}, {0x70,0x40}, {0x70,0xc0},
{0x73,0x40}, {0x73,0xc0}, {0x72,0x40}, {0x72,0xc0},
{0x2a,0x00}, {0x2e,0x00}, {0x22,0x00}, {0x26,0x00},
{0x3a,0x00}, {0x3e,0x00}, {0x32,0x00}, {0x36,0x00},
{0x0a,0x00}, {0x0e,0x00}, {0x02,0x00}, {0x06,0x00},
{0x1a,0x00}, {0x1e,0x00}, {0x12,0x00}, {0x16,0x00},
{0x55,0x00}, {0x57,0x00}, {0x51,0x00}, {0x53,0x00},
{0x5d,0x00}, {0x5f,0x00}, {0x59,0x00}, {0x5b,0x00},
{0x45,0x00}, {0x47,0x00}, {0x41,0x00}, {0x43,0x00},
{0x4d,0x00}, {0x4f,0x00}, {0x49,0x00}, {0x4b,0x00},
{0x7e,0xa8}, {0x7e,0xb8}, {0x7e,0x88}, {0x7e,0x98},
{0x7e,0xe8}, {0x7e,0xf8}, {0x7e,0xc8}, {0x7e,0xd8},
{0x7e,0x28}, {0x7e,0x38}, {0x7e,0x08}, {0x7e,0x18},
{0x7e,0x68}, {0x7e,0x78}, {0x7e,0x48}, {0x7e,0x58},
{0x7f,0xa8}, {0x7f,0xb8}, {0x7f,0x88}, {0x7f,0x98},
{0x7f,0xe8}, {0x7f,0xf8}, {0x7f,0xc8}, {0x7f,0xd8},
{0x7f,0x28}, {0x7f,0x38}, {0x7f,0x08}, {0x7f,0x18},
{0x7f,0x68}, {0x7f,0x78}, {0x7f,0x48}, {0x7f,0x58},
{0x7a,0xa0}, {0x7a,0xe0}, {0x7a,0x20}, {0x7a,0x60},
{0x7b,0xa0}, {0x7b,0xe0}, {0x7b,0x20}, {0x7b,0x60},
{0x78,0xa0}, {0x78,0xe0}, {0x78,0x20}, {0x78,0x60},
{0x79,0xa0}, {0x79,0xe0}, {0x79,0x20}, {0x79,0x60},
{0x7d,0x50}, {0x7d,0x70}, {0x7d,0x10}, {0x7d,0x30},
{0x7d,0xd0}, {0x7d,0xf0}, {0x7d,0x90}, {0x7d,0xb0},
{0x7c,0x50}, {0x7c,0x70}, {0x7c,0x10}, {0x7c,0x30},
{0x7c,0xd0}, {0x7c,0xf0}, {0x7c,0x90}, {0x7c,0xb0},
{0x95,0x80}, {0x94,0x80}, {0x97,0x80}, {0x96,0x80},
{0x91,0x80}, {0x90,0x80}, {0x93,0x80}, {0x92,0x80},
{0x9d,0x80}, {0x9c,0x80}, {0x9f,0x80}, {0x9e,0x80},
{0x99,0x80}, {0x98,0x80}, {0x9b,0x80}, {0x9a,0x80},
{0x8a,0xc0}, {0x8a,0x40}, {0x8b,0xc0}, {0x8b,0x40},
{0x88,0xc0}, {0x88,0x40}, {0x89,0xc0}, {0x89,0x40},
{0x8e,0xc0}, {0x8e,0x40}, {0x8f,0xc0}, {0x8f,0x40},
{0x8c,0xc0}, {0x8c,0x40}, {0x8d,0xc0}, {0x8d,0x40},
{0xd6,0x00}, {0xd2,0x00}, {0xde,0x00}, {0xda,0x00},
{0xc6,0x00}, {0xc2,0x00}, {0xce,0x00}, {0xca,0x00},
{0xf6,0x00}, {0xf2,0x00}, {0xfe,0x00}, {0xfa,0x00},
{0xe6,0x00}, {0xe2,0x00}, {0xee,0x00}, {0xea,0x00},
{0xab,0x00}, {0xa9,0x00}, {0xaf,0x00}, {0xad,0x00},
{0xa3,0x00}, {0xa1,0x00}, {0xa7,0x00}, {0xa5,0x00},
{0xbb,0x00}, {0xb9,0x00}, {0xbf,0x00}, {0xbd,0x00},
{0xb3,0x00}, {0xb1,0x00}, {0xb7,0x00}, {0xb5,0x00},
{0x81,0x58}, {0x81,0x48}, {0x81,0x78}, {0x81,0x68},
{0x81,0x18}, {0x81,0x08}, {0x81,0x38}, {0x81,0x28},
{0x81,0xd8}, {0x81,0xc8}, {0x81,0xf8}, {0x81,0xe8},
{0x81,0x98}, {0x81,0x88}, {0x81,0xb8}, {0x81,0xa8},
{0x80,0x58}, {0x80,0x48}, {0x80,0x78}, {0x80,0x68},
{0x80,0x18}, {0x80,0x08}, {0x80,0x38}, {0x80,0x28},
{0x80,0xd8}, {0x80,0xc8}, {0x80,0xf8}, {0x80,0xe8},
{0x80,0x98}, {0x80,0x88}, {0x80,0xb8}, {0x80,0xa8},
{0x85,0x60}, {0x85,0x20}, {0x85,0xe0}, {0x85,0xa0},
{0x84,0x60}, {0x84,0x20}, {0x84,0xe0}, {0x84,0xa0},
{0x87,0x60}, {0x87,0x20}, {0x87,0xe0}, {0x87,0xa0},
{0x86,0x60}, {0x86,0x20}, {0x86,0xe0}, {0x86,0xa0},
{0x82,0xb0}, {0x82,0x90}, {0x82,0xf0}, {0x82,0xd0},
{0x82,0x30}, {0x82,0x10}, {0x82,0x70}, {0x82,0x50},
{0x83,0xb0}, {0x83,0x90}, {0x83,0xf0}, {0x83,0xd0},
{0x83,0x30}, {0x83,0x10}, {0x83,0x70}, {0x83,0x50},
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
static const uint8_t lintoalaw[256] = {
Audio changes: - Change the way attach and open works to allow multiple audio devices. - Split the mulaw.c file into two to avoid dragging in mulaw convertsion when they are not needed. Add 16 bit alaw/mulaw tables. - Change the way audio properties are gotten. - Recognize more versions os SoundBlaster.
1997-08-01 02:33:08 +04:00
0x2a, 0x2a, 0x2a, 0x2a, 0x2b, 0x2b, 0x2b, 0x2b,
0x28, 0x28, 0x28, 0x28, 0x29, 0x29, 0x29, 0x29,
0x2e, 0x2e, 0x2e, 0x2e, 0x2f, 0x2f, 0x2f, 0x2f,
0x2c, 0x2c, 0x2c, 0x2c, 0x2d, 0x2d, 0x2d, 0x2d,
0x22, 0x22, 0x22, 0x22, 0x23, 0x23, 0x23, 0x23,
0x20, 0x20, 0x20, 0x20, 0x21, 0x21, 0x21, 0x21,
0x26, 0x26, 0x26, 0x26, 0x27, 0x27, 0x27, 0x27,
0x24, 0x24, 0x24, 0x24, 0x25, 0x25, 0x25, 0x25,
0x3a, 0x3a, 0x3b, 0x3b, 0x38, 0x38, 0x39, 0x39,
0x3e, 0x3e, 0x3f, 0x3f, 0x3c, 0x3c, 0x3d, 0x3d,
0x32, 0x32, 0x33, 0x33, 0x30, 0x30, 0x31, 0x31,
0x36, 0x36, 0x37, 0x37, 0x34, 0x34, 0x35, 0x35,
0x0a, 0x0b, 0x08, 0x09, 0x0e, 0x0f, 0x0c, 0x0d,
0x02, 0x03, 0x00, 0x01, 0x06, 0x07, 0x04, 0x05,
0x1a, 0x18, 0x1e, 0x1c, 0x12, 0x10, 0x16, 0x14,
0x6a, 0x6e, 0x62, 0x66, 0x7a, 0x72, 0x4a, 0x5a,
0xd5, 0xc5, 0xf5, 0xfd, 0xe5, 0xe1, 0xed, 0xe9,
0x95, 0x97, 0x91, 0x93, 0x9d, 0x9f, 0x99, 0x9b,
0x85, 0x84, 0x87, 0x86, 0x81, 0x80, 0x83, 0x82,
0x8d, 0x8c, 0x8f, 0x8e, 0x89, 0x88, 0x8b, 0x8a,
0xb5, 0xb5, 0xb4, 0xb4, 0xb7, 0xb7, 0xb6, 0xb6,
0xb1, 0xb1, 0xb0, 0xb0, 0xb3, 0xb3, 0xb2, 0xb2,
0xbd, 0xbd, 0xbc, 0xbc, 0xbf, 0xbf, 0xbe, 0xbe,
0xb9, 0xb9, 0xb8, 0xb8, 0xbb, 0xbb, 0xba, 0xba,
0xa5, 0xa5, 0xa5, 0xa5, 0xa4, 0xa4, 0xa4, 0xa4,
0xa7, 0xa7, 0xa7, 0xa7, 0xa6, 0xa6, 0xa6, 0xa6,
0xa1, 0xa1, 0xa1, 0xa1, 0xa0, 0xa0, 0xa0, 0xa0,
0xa3, 0xa3, 0xa3, 0xa3, 0xa2, 0xa2, 0xa2, 0xa2,
0xad, 0xad, 0xad, 0xad, 0xac, 0xac, 0xac, 0xac,
0xaf, 0xaf, 0xaf, 0xaf, 0xae, 0xae, 0xae, 0xae,
0xa9, 0xa9, 0xa9, 0xa9, 0xa8, 0xa8, 0xa8, 0xa8,
0xab, 0xab, 0xab, 0xab, 0xaa, 0xaa, 0xaa, 0xaa,
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
};
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
#define DEFINE_FILTER(name) \
static int \
name##_fetch_to(stream_fetcher_t *, audio_stream_t *, int); \
stream_filter_t * \
name(struct audio_softc *sc, const audio_params_t *from, \
const audio_params_t *to) \
{ \
DPRINTF(("Construct '%s' filter.\n", __func__)); \
return auconv_nocontext_filter_factory(name##_fetch_to); \
} \
static int \
name##_fetch_to(stream_fetcher_t *self, audio_stream_t *dst, int max_used)
DEFINE_FILTER(mulaw_to_linear8)
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (dst->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = mulawtolin16[*s][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = mulawtolin16[*s][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Change the interface between high and lowlevel audio drivers again: Set the encoding parameters slightly differently. Remove the SW encoding/decodinf functions from this interface and move them to the audio_parameter struct; this is both more efficient and flexible.
1997-05-10 02:16:27 +04:00
}
GUS audio driver from John Kohl.
1995-07-19 23:58:09 +04:00
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(mulaw_to_linear16)
Add alaw/mulaw conversion to/from 8 bit signed linear.
1998-05-22 17:05:31 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
if ((err = this->prev->fetch_to(this->prev, this->src, max_used / 2)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
switch (dst->param.encoding) {
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][1];
d[1] = mulawtolin16[s[0]][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][0];
d[1] = mulawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][1];
d[1] = mulawtolin16[s[0]][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = mulawtolin16[s[0]][0] ^ 0x80;
d[1] = mulawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
Add alaw/mulaw conversion to/from 8 bit signed linear.
1998-05-22 17:05:31 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Add alaw/mulaw conversion to/from 8 bit signed linear.
1998-05-22 17:05:31 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear16_to_mulaw)
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used * 2)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
switch (this->src->param.encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[1] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[0] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[1]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintomulaw[s[0]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear8_to_mulaw)
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (this->src->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintomulaw[*s];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else { /* SLINEAR_LE */
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintomulaw[*s ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Make little endian and big endian version of all functions that deal with 16 bit samples. Let the drivers decide which ones to use.
1999-11-01 21:12:19 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(alaw_to_linear8)
Change the interface between high and lowlevel audio drivers again: Set the encoding parameters slightly differently. Remove the SW encoding/decodinf functions from this interface and move them to the audio_parameter struct; this is both more efficient and flexible.
1997-05-10 02:16:27 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (dst->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = alawtolin16[*s][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else { /* SLINEAR */
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = alawtolin16[*s][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(alaw_to_linear16)
Add mulaw_to_slinear16 and alaw_to_slinear16.
1998-08-09 23:22:15 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
max_used = (max_used + 1) & ~1; /* round up to even */
if ((err = this->prev->fetch_to(this->prev, this->src, max_used / 2)))
return err;
m = (dst->end - dst->start) & ~1;
m = min(m, max_used);
switch (dst->param.encoding) {
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][1];
d[1] = alawtolin16[s[0]][0];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][0];
d[1] = alawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][1];
d[1] = alawtolin16[s[0]][0] ^ 0x80;
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 2, m) {
d[0] = alawtolin16[s[0]][0] ^ 0x80;
d[1] = alawtolin16[s[0]][1];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
Add mulaw_to_slinear16 and alaw_to_slinear16.
1998-08-09 23:22:15 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
Add mulaw_to_slinear16 and alaw_to_slinear16.
1998-08-09 23:22:15 +04:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear8_to_alaw)
Add slinear16_to_mulaw_le().
2002-02-10 09:27:06 +03:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
if (this->src->param.encoding == AUDIO_ENCODING_ULINEAR_LE) {
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintoalaw[*s];
} FILTER_LOOP_EPILOGUE(this->src, dst);
} else { /* SLINEAR_LE */
FILTER_LOOP_PROLOGUE(this->src, 1, dst, 1, m) {
*d = lintoalaw[*s ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
DEFINE_FILTER(linear16_to_alaw)
Add Alaw encoding emulation.
1997-05-28 04:07:46 +04:00
{
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
stream_filter_t *this;
int m, err;
this = (stream_filter_t *)self;
if ((err = this->prev->fetch_to(this->prev, this->src, max_used * 2)))
return err;
m = dst->end - dst->start;
m = min(m, max_used);
switch (this->src->param.encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[1] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_SLINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[0] ^ 0x80];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_LE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[1]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
case AUDIO_ENCODING_ULINEAR_BE:
FILTER_LOOP_PROLOGUE(this->src, 2, dst, 1, m) {
d[0] = lintoalaw[s[0]];
} FILTER_LOOP_EPILOGUE(this->src, dst);
break;
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}
merge kent-audio1 branch, which introduces audio filter pipeline to the MI audio framework Summary of changes: * struct audio_params - remove sw_code, factor, factor_denom, hw_sample_rate, hw_encoding ,hw_precision, and hw_channels. Conversion information is conveyed by stream_filter_list_t. - change the type of sample_rate: u_long -> u_int - add `validbits,' which represents the valid data size in precision bits. It is required in order to distinguish 24/32bit from 24/24bit or 32/32bit. * audio_hw_if - add two parameters to set_params() stream_filter_list_t *pfil, stream_filter_list *rfil A HW driver should set filter recipes for requested formats - constify audio_params parameters of trigger_output() and trigger_input(). They represent audio formats for the hardware. - make open() and close() optional - add int (AUMODE_PLAY or AUMODE_RECORD) and audio_params_t parameters to round_blocksize() * sw_code is replaced with stream_filter_t. stream_filer_t converts audio data in an input buffer and writes into another output buffer unlike sw_code, which converts data in single buffer. converters in dev/auconv.c, dev/mulaw.c, dev/aurateconv.c, dev/tc/bba.c, dev/ic/msm6258.c, and arch/arm/iomd/vidcaudio.c are reimplemented as stream_filter_t * MI audio - audiosetinfo() builds filter pipelines from stream_filter_list_t filled by audio_hw_if::set_params() - audiosetinfo() returns with EINVAL if mmapped and set_params() requests filters - audio_write(), audio_pint(), and audio_rint() invoke a filter pipeline. - ioctl() for FIONREAD, AUDIO_WSEEK, AUDIO_GETIOFFS, AUDIO_GETOOFFS, and audio_prinfo::{seek,samples} for AUDIO_GETINFO handle values for a buffer nearest to userland. * add `struct device *' parameter to ac97_attach() * all of audio HW drivers follow audio_hw_if and ac97 changes
2005-01-11 01:01:36 +03:00
return 0;
add the follwings: - ulinear16_to_mulaw_le() - ulinear16_to_mulaw_be() - slinear16_to_mulaw_be() - ulinear16_to_alaw_le() - ulinear16_to_alaw_be()
2004-11-05 19:31:14 +03:00
}